This invention relates generally to techniques for coding sound signals, which are originally analog signals, and more particularly, the present invention relates to method and apparatus for compressively approximating an input sound signal waveform to produce an output digital signal, so that transmission of and/or reproduction from the digital signal can be readily performed after the amound of data has been reduced.
Up to this time, various measures have been taken for reducing the amount of digital data or information indicative of original input sound signal waveform so that transmission and/or storage of the digital data can be effectively performed at low cost. In order to reduce or compress the amount of digital data, in the prior art, for instance, the amplitude of the original sound signal is logarithmitically compressed (log PCM), a difference between adjacent data is obtained (DPCM), or delta modulation is performed, as is well known.
In any of the above-mentioned conventional measures for reducing data amount, the interval between adjacent sampling pulses, i.e. the sampling period, is determined to be less than a reciprocal of a frequency which is twice the maximum frequency of the objective signal to be coded or sampled, so that aliasing distortion or noise is prevented from occurring. However, it is well known that occurrence of so called quantization distortion or noise is inevitable in theory when coding an analog signal into a digital signal which assumes finite number of discrete values. Because of the presence of the quantization noise distributed in the frequency range of an analog signal which has been restored from the digital signal, the quality of the restored analog signal has been deteriorated.
The common feature in conventional coding or sampling techniques is that a net or mesh extending in both directions of amplitude and time is set to pick up or sample to indicate each value close to each knot of the mesh. Therefore, even though the mesh is made fine and precise in only the direction of time, the error between the original signal waveform and the restored signal waveform does not necessarily become minimum. In other words, when care is taken in connection with only time axis in accordance with sampling theorem so as to prevent aliasing noises, quantization distortion does exist, where the degree of the quantization distortion is determined by the fineness of the mesh in the direction of amplitude.
In the above-mentioned conventional techniques, the sampling mesh is made fine in the direction of time axis so as to ensure adequate frequency range and to prevent aliasing noises. For instance, the sampling rate or frequency per one second is 8000 for vocal sounds, and 44000 for music. On the other hand, the fineness in the direction of amplitude is determined by the number of bits which represent each sampled value, and it is required to provide 7 to 8 bits for vocal sounds, and 12 to 16 bits for music.
Consequently, the amount of basic or original data or information per a second is as large as 56000 to 704000 bits. In the prior art, this enormous amount of data has been reduced in various ways as described in the above. However, the entire amount of data could only be reduced by half according to the conventional techniques. It is obvious that such a large amount of data is not economical since it takes a relatively long period of time for transmitting such a large amount of data, while a large storage device is required for storing the same. Furthermore, it takes a relatively long period of time in finding a given data from such a large amount of data.